1. Field of the Invention
The present invention relates to an adaptive active noise cancellation apparatus and, more particularly, an adaptive active noise cancellation apparatus including an adaptive control system capable of adaptively obtaining a filter coefficient used for an active noise cancellation control system in a state wherein a sound source is continuously driven.
2. Description of the Related Art
Recently, an active noise cancellation apparatus based on an acoustic control technique has been developed. In this active noise cancellation apparatus, in general, a noise generated by a primary noise source is detected by a sensor, and a sound generator such as speaker is operated in response to a signal obtained by filtering a signal from the sensor through a filter having a predetermined filter coefficient, thereby actively cancelling the noise at a control target point by a sound generated by the sound generator. The principle of such noise cancellation is disclosed in U.S. Pat. No. 2,043,416.
In such an active noise cancellation apparatus, a filter coefficient required for noise cancellation is obtained by using the principle of a digital filter. More specifically, if a transfer function in a spatial system is represented by H(.omega.); and a signal input to a space, X(.omega.), an output Y(.omega.) in a frequency region is given by EQU Y(.omega.)=H(.omega.).multidot.X(.omega.) (1)
However, an output in a time domain is represented by convolution integration: ##EQU1## where h(t) is the impulse response. In the embodiment, the frequency domain is represented by a large letter such as Y, H, X, S, G, M, L, E, etc., while the time domain is indicated by a small letter such as y, h, x, s, g, m, l, e, etc.
As is apparent from equation (2), the output represented by a product in the frequency region is obtained from the sum of products in the time domain, i.e., multiplying the impulse response and values obtained by sequentially delaying an input value in the time domain by .tau., and adding the resultant products together. That is, an operation equivalent to equation (1) can be realized by a product summation operation and a delay circuit having a delay time .tau.. In an actual control operation or the like, the range of integration is finite, and a corresponding arithmetic operation is generally executed in a digital manner. Therefore, an equation corresponding to equation (2) is ##EQU2##
This is generally called an FIR (Finite Impulse Response) filter. In equation (3), h(k) is the impulse response, i.e., the filter coefficient of this filter. In an active noise cancellation apparatus, an impulse response, i.e., a filter coefficient, used for noise cancellation control must be obtained in advance. A method of obtaining a filter coefficient will be described below with reference to FIG. 1. FIG. 1 shows a case wherein an active noise cancellation apparatus 4 prevents a noise generated by a noise source 2 housed in a duct 1 from leaking through an opening portion 3 of the duct 1. A sensor, e.g., an acceleration pickup 5 for detecting vibrations, detects a noise generated by the noise source 2 by using another signal having a high correlation with this noise. A filter coefficient required to constitute an FIR filter is set in a signal processor 6. A speaker 7 generates an active sound required for noise cancellation. An evaluation microphone 8 is arranged to evaluate a cancellation effect at a noise cancellation target point.
Assuming that a transfer function between the noise source 2 and the evaluation microphone 8 is represented by L; a transfer function between the speaker 7 and the evaluation microphone 8, M; and an noise signal generated by the noise source 2 (and detected by the acceleration pickup 5), S, a signal I observed by the evaluation microphone 8 is given by EQU I=S.multidot.L+S.multidot.G.multidot.M (4)
where G is the transfer function required for noise cancellation. When the noise is completely canceled at the noise cancellation target point, the value I in equation (4) is given by I=0. Therefore, the transfer function G must be given by EQU G=-L/M (5)
Equation (5) is normally calculated by a fast Fourier transform in a frequency region. An impulse response is obtained by an inverse Fourier transform of the resulting value. The obtained impulse response is set in the signal processor 6 as a filter coefficient.
The active noise cancellation apparatus 4 having the above-described arrangement, however, cannot cope with a generated noise by using the fixed filter coefficient obtained from equation (5) when a transfer function in a spatial system for a space changes in quality over time, or the characteristics (e.g., correlation) of the noise source change.
In order to cope with the above inconvenience, therefore, an adaptive active noise cancellation apparatus using an adaptive control technique has recently been developed (disclosed in, e.g., "Study of Electronic Sound Cancellation System for Piping: Adaptive Type DSM System", Lecture Papers of Japanese Association of Acoustics, pp. 367-368). Adaptive type active noise cancellation apparatuses of various schemes are available. According to the most simple apparatus, the signal processor 6 functions as an adaptive controller and, for example, every time the output I from the evaluation microphone 8 exceeds a predetermined level, the transfer function G with which the output I from the evaluation microphone 8 is minimized is obtained, and the filter coefficient in the signal processor 6 is adaptively updated. That is, in this adaptive type active noise cancellation apparatus, when an active noise is output from the speaker 7 upon a multiplication of a signal S and a filter coefficient, the transfer function G with which a sound obtained by synthesizing the active sound and the noise sound from the noise source 2 becomes zero at the position of the evaluation microphone 8 is obtained, and an impulse response, i.e., a filter coefficient, is obtained from this transfer function G. In the adaptive type active noise cancellation apparatus having such an arrangement, since a filter coefficient can be adaptively obtained while a continuous operation of the noise source 2 is allowed, only few limitations are imposed on the noise source 2, and the overall arrangement of the apparatus can be simplified.
In the adaptive type active noise cancellation apparatus having the above-described arrangement, however, the following problems are posed. FIG. 2 shows an equivalent circuit diagram of an adaptive control system in the adaptive type active noise cancellation apparatus having the above arrangement. Referring to FIG. 2, reference symbol M denotes a transfer function between a speaker 7 and an evaluation microphone 8; L, a transfer function between the noise source 2 and the evaluation microphone 8; and e, an error signal observed by the evaluation microphone 8. The transfer function G is determined so as to set the error signal e to be zero. However, as is apparent from the arrangement shown in FIG. 2, since adaptive control is performed while the error signal e includes the influences of the transfer function M in the adaptive control system incorporated in the conventional apparatus, the adaptive control system does not operate to set the signal e to be zero. More specifically, one element, i.e., g.sub.new,1, of a new filter coefficient g.sub.new (impulse response) obtained in the arrangement shown in FIG. 1 is given by ##EQU3## where a small letter indicates a time domain, and a bold letter indicates a column vector. The apparatus shown in FIG. 1 does not execute calculations of ##EQU4## For this reason, in the adaptive controller shown in FIG. 1, the filter coefficient cannot be converged to a desired value. Therefore, in the adaptive active noise cancellation apparatus incorporating the adaptive control system shown in FIG. 1, a good noise cancellation effect cannot be obtained. As described above, in the conventional adaptive active noise cancellation apparatus having the function of adaptively updating the filter coefficient in a state wherein continuous driving of a noise source is allowed, the convergence of the filter coefficient is interfered by the influences of the transfer function included in an error signal. Therefore, proper adaptive control cannot be realized.